The following paragraphs provide some background and prior art information in order to illustrate the specific characteristics of noise canceling microphones that are improved by this invention. Prior inventions have failed to provide for design robustness and the wide noise suppression bandwidth required for clear communication in high ambient noise fields. This invention focuses on providing a new noise canceling microphone using controller and algorithmic features that drastically improve the performance over the prior art noise canceling microphones. This application relies on the provisional Patent Application Ser. No. 60/242,952 filed Oct. 24, 2000, with inventors Michael Vaudrey and William Saunders entitled “Improved Noise Canceling Microphone”.
To review the nominal field being considered, it is recalled that passive noise canceling microphones typically incorporate a single membrane to sense ambient sound, where the housing of that membrane is open to the environment on both sides. Far-field sounds impact the membrane (essentially) equally on both sides, generating no net movement, and thus a low sensitivity. Near field sounds (such as when the microphone is placed close to a speaker's mouth) cause the membrane to move more significantly in one direction than another, causing a higher sensitivity. This higher sensitivity to close-range voice versus lower sensitivity to far-field ambient noise, provides a low frequency improvement in the signal-to-noise ratio because of the associated far field noise rejection; thus improving low frequency speech intelligibility. There are a multitude of patents that cover the passive noise canceling microphone concept in various ways including: U.S. Pat. Nos. 4,258,235, 3,995,124, 5,329,593, and 5,511,130 among others. The microphone invention described here is an active microphone and is therefore different from this prior art regarding passive elements.
A second category of noise canceling microphones will be referred to as active noise canceling microphones. The most rudimentary active noise canceling microphones perform identically to the passive noise canceling microphones mentioned above. The structural difference is that an active element such as a subtraction circuit is employed in order to electronically difference two microphone signals, in order to generate the noise canceled output signal. The two microphones are positioned facing away from each other, where one is directed toward the desired signal source, or speaker's mouth. There are patents focusing on the use of active elements in creating a noise canceling microphone including U.S. Pat. Nos. 5,303,307 and 5,511,130. The algorithms and design features presented herein are not anticipated by any of this prior art.
More advanced implementations of noise canceling microphones have arisen as a result of increased DSP processing capabilities, the present invention included. These adaptive active noise cancellation microphones typically include the use of an adaptive filter as part of the active canceling element and provide improved performance over both the passive and active noise canceling microphones. The invention disclosed herein is significantly different from the prior art in this area as evidenced below.
U.S. Pat. No. 5,917,921 by Sasaki et. al. is a very general embodiment of an adaptive active noise canceling microphone. Sasaki uses an adaptive filter with two microphone signals to reduce the noise in one of those signals, using the other as a reference input to the adaptive filter. The inventive elements described in the present invention are not described or anticipated by the disclosure of Sasaki, which only focuses on the general idea of using an adaptive filter with two microphones for the purpose of reducing wind noise. The specific embodiments described by this invention are not anticipated by Sasaki.
U.S. Pat. No. 5,953,380 by Ikeda focuses on a very specific method for controlling the convergence parameter of the adaptation process as a function of the two input signals. A complex series of delays and power estimations creates a single convergence parameter for the time domain adaptive filter. This single convergence parameter is varied with the detection of the speech, as determined by the “SN power ratio estimation”. Ikeda does not anticipate the present inventions because the need for robust performance in a physical product is not discussed; nor does Ikeda anticipate the concept of multiple frequency-dependent convergence parameters or the use of frequency-domain adaptive control.
U.S. Pat. No. 5,978,824, also by Ikeda, is an adaptive filtering method for creating a “clean” estimate of the noise as well as a “clean” estimate of the desired signal. The two adaptive filters create estimates of the desired signal and the noise signal, which are independently used to generate convergence parameters for the two adaptive filters. The two adaptive filters used in Ikeda's invention are used to a) generate a more accurate estimate of the signal to noise at any given time and b) create more accurate estimates of the speech, as well as the noise. The two adaptive filters used in the present invention provide an entirely different effect focused on improving robustness during quickly changing ambient noise disturbances; in addition, the arrangement of the adaptive filters in the present invention is completely different from and is not anticipated by Ikeda.
U.S. Pat. No. 5,473,684 by Bartlett and Zuniga describes two first-order differential microphones that are used to create an adaptive second-order differential microphone. The present invention uses two omni-directional microphones to create a single, adaptive, first-order differential microphone. The use of omni-directional microphones simplifies the physical construction of the microphone assembly, since both transducer backplanes can remain secured in the housing. (FOD microphones must be open on both sides in order to be effective). No mention by Bartlett is made concerning the use of two adaptive filters for optimizing the robust control of ambient noise. In addition, no mention is made of using a frequency domain adaptive algorithm for controlling multiple convergence parameters of individual frequencies.
U.S. Pat. No. 5,473,702 by Yoshida et al. controls the adaptation of the adaptive noise-canceling filter by adjusting the convergence parameter as a function of the error signal. There are several options that are discussed through a complex rule-based system that ultimately decides when the algorithm should temporarily cease adaptation. Frequency domain control of adaptation is not anticipated by Yoshida, nor is the use of two adaptive filters for robust performance of a two-element adaptive noise canceling microphone design.
Finally, U.S. Pat. No. 5,319,736 by Hunt describes a digital signal processing system that creates a frequency spectrum of speech from noisy speech to be used by a speech recognition system. This system does not anticipate using multiple adaptive filters as disclosed herein. In addition, Hunt's system does not anticipate performing real-time frequency domain adaptive filtering for communication microphone applications. Instead the output of his system is used as an input to a frequency domain vocoder.
In summary, this review of the prior art in adaptive noise canceling microphones directly points to the need for a more robust design of an adaptive noise-canceling microphone where the minimum performance is at least as good as passive noise canceling microphones at all times and the maximum performance can far exceed that of the existing noise canceling microphones. Tests have shown that in highly reverberant environments, the passive noise control microphone design can perform better than the prior art adaptive noise canceling microphones discussed above, if safeguards are not applied. The dual-filter embodiment of this invention disclosed herein is such a safeguard that ensures the adaptive noise canceling microphone will always perform at least as well as the passive version, thereby improving the robustness of any noise canceling microphone previously described in the prior art.
The second failing of the prior adaptive noise canceling microphone designs is that fast variations in the noise field cannot be tracked when the adaptive filter has a small convergence coefficient. This problem leads to increased average noise levels for the adaptive filter arrangements discussed by others. The first-stage, single-weight adaptive filter of the present invention eliminates the degradation associated with fast tracking of noise field variations.
Finally, the prior art does not anticipate the need for frequency domain adaptation. This is a problem for all of those previously discussed inventions because the adaptation of the entire filter is halted at every frequency every time there is a component of speech detected. This leads to sub-optimal wideband noise suppression. The solution offered by the present invention is to only adapt individual frequency bins, allowing non-speech, noise frequencies to be adapted while simultaneously halting adaptation for those frequency bins dominated by speech content. Detailed descriptions of the invention are provided next.